[asterisk-ss7] libss7 Audio after DTMF
Attila Domjan
adomjan at tvnet.hu
Wed Apr 21 07:54:50 CDT 2010
Hi,
check, if not exists add it!
On Wed, 2010-04-21 at 08:36 -0400, Dave George wrote:
> Hi Atilla,
>
> I had this issue during setup and I got this from the list:
>
>
> " Hi, check the existence of the
>
> p->dialing = 0;
>
> in chan_dahdi.c, static void *ss7_linkset(void *data) after the case ISUP_EVENT_CON:
> case ISUP_EVENT_ANM:
>
> and
> case ISUP_EVENT_CPG:
>
> near p->progress = 1; "
>
>
> Now that I have traffic on the issue is back on 30% of the calls. Do I have to add it after case ISUP_EVENT_ACM:, or should the above solve the problem.
>
>
>
> Thanks,
> Dave George
> Teletone Inc.
> 561 674 3838
>
>
> -----Original Message-----
> From: asterisk-ss7-bounces at lists.digium.com [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Attila Domjan
> Sent: Wednesday, April 21, 2010 3:58 AM
> To: asterisk-ss7 at lists.digium.com
> Subject: Re: [asterisk-ss7] libss7 Audio after DTMF
>
> I think it is the bug what I wrote it many times to this list, the missing
>
> p->proceeding = 1;
> p->dialing = 0;
>
> after the case ISUP_EVENT_ACM:, in static void *ss7_linkset(void *data)
>
> A
>
> On Wed, 2010-04-21 at 01:47 -0300, Rafael Prado Rocchi wrote:
> > What asterisk version are you using?
> > There's a bug like it in some asterisk versions where you have to
> > press a key before hearing the audio.
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: asterisk-ss7-bounces at lists.digium.com [mailto:asterisk-ss7-
> > > bounces at lists.digium.com] On Behalf Of Dave George
> > > Sent: terça-feira, 20 de abril de 2010 10:34
> > > To: asterisk-ss7 at lists.digium.com
> > > Subject: [asterisk-ss7] libss7 Audio after DTMF
> > >
> > > I am using libss7 on an ansi ss7 interconnect. I have two T1s on a
> > > Digium TE410P card. On many of the calls I have to hit a key before
> > > hearing any audio. Any suggestions welcome. Happens about 20 % of
> > > the calls.
> > >
> > >
> > > System.conf
> > > span=1,1,0,esf,b8zs
> > > span=2,0,0,esf,b8zs
> > > span=3,0,0,esf,b8zs
> > > span=4,2,0,esf,b8zs
> > > mtp2=1
> > > bchan=2-24
> > > mtp2=73
> > > bchan=74-96
> > >
> > >
> > >
> > > chan_dahdi.conf
> > >
> > > ; All settings apply to linkset 1
> > > linkset = 1
> > > pointcode = x-x-x
> > > adjpointcode = x-x-x
> > > defaultdpc = x-x-x
> > >
> > > slc=0
> > > sigchan = 1
> > > cicbeginswith = 102
> > > channel = 2-24
> > >
> > > slc=1
> > > sigchan = 73
> > > cicbeginswith = 126
> > > channel = 74-96
> > >
> > >
> > >
> > > Thanks,
> > > Dave George
> > > 561 674 3838
> > >
> > >
> > >
> > >
> > > --
> > > ____________________________________________________________________
> > > _
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> > > --
> > >
> > > asterisk-ss7 mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-ss7
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 190 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100421/efae6a57/attachment-0001.pgp
More information about the asterisk-ss7
mailing list